In the past, single-channel as well as multi-channel audio amplification systems have been devised to compress the dynamic range of audio signals. However, both types of systems suffer from different, mutually exclusive limitations. Single-channel systems preserve spectral contrast but cannot provide adequate frequency-dependent compressive gain. In addition, such systems unnecessarily suppress or distort signal information in situations with low signal-to-noise ratios, where strong interfering components in remote frequency regions can control the gain. For the same reason, steady background sounds can acquire an objectionable modulation in the presence of fluctuating foreground sounds. Multi-channel systems, on the other hand, can provide frequency-dependent compression and can ensure audibility of weak signal components in the presence of wideband interferers if these systems are sufficiently fast-acting. However, by reducing spectral contrast across channels, they diminish spectral pattern information.
One place where this is observed is in hearing correction devices, such as hearing aids. Persons with sensorineural hearing loss experience reduced sensitivity to faint, low-level sounds and loudness recruitment, i.e., an abnormally steep growth of perceived loudness with sound level. In addition, due to the partial loss of frequency-dependent compressive gain in the impaired auditory system, the level-dependent auditory frequency tuning is affected. Compared to the normally-functioning auditory system, the tuning is particularly degraded at low sound levels resulting in a more static tuning as a function of level. One goal of assistive technology is to compensate for these consequences of sensorineural hearing loss, in order to improve perceived sound quality and aided performance of hearing-impaired listeners on advanced auditory functions such as speech or music perception in complex auditory environments. However, conventional single-channel and multi-channel systems suffer from the aforementioned problems, which sometimes can even compound the difficulties experienced by hearing-impaired listeners. The reduction of spectral contrast by multi-channel systems, for example, will only exacerbate the challenges faced by the impaired auditory system with its degraded frequency resolution.
FIG. 1 illustrates a basic prior art compression system. In the first stage, the incoming signal is buffered and spectrally analyzed, for example by using an FFT, warped FFT, or a time-domain filter-bank analysis (e.g., Kates, J. M., 2008, “Digital Hearing Aids,” Plural Publishing, San Diego, Calif.). Next, typically the signal power or signal envelope (for brevity, only signal power is referred to in the following) in each band is estimated by a power detector and smoothed by a power integrator which informs the subsequent gain calculation (throughout this application, “band” refers to static spectral bands). This gain is then applied to the individual band signals and the overall signal is re-synthesized by using an inverse FFT or a synthesis filter bank in conjunction with overlap add synthesis. FIG. 2 shows an alternative prior art implementation where the compressive-gain calculation is “side-branched”, with the compressive-gain filter transformed into the time domain and applied via time-domain convolution.
Static hybrid systems such as the one devised by White in U.S. Pat. No. 4,701,953 entitled “Signal Compression System” (1987), use broadly overlapping analysis filters for envelope/power detection and narrow synthesis filters, preserve spectral contrast and provide frequency-dependent gain functions, but still fail to provide adequate signal gain in situations with low signal-to-noise ratios.
What is needed in the art is a way to provide level-dependent processing of sounds that optimizes both spectral contrast and gain.